sip – yesterday, today, & tomorrow
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DESCRIPTIONSIP Yesterday, Today, & Tomorrow. Jon Murphy Sr. Network Application Engineer tw telecom. Introduction. Jon Murphy Sr. Network Application Engineer tw telecom firstname.lastname@example.org (614) 255-2132 (office) (614) 313-6925 (cell). GOAL/Agenda. - PowerPoint PPT Presentation
SIP Yesterday, Today, & Tomorrow
Jon MurphySr. Network Application Engineertw telecom
IntroductionJon MurphySr. Network Application Engineertw email@example.com(614) 255-2132 (office)(614) 313-6925 (cell)
GOAL/AgendaI hope you leave here today understanding:
What is SIP? Overall Concept, Definition, and Components.
How did SIP get here? History of SIP/VOIP
What does the future of SIP/VOIP look like?
A little history, a little overview, a little tech, a little bit of everything
Warning No Commercials
Lets build a SIP hamburger with minimal bun!
I will try to add some spice with pickles and tomatoes but at the end of the day this is still SIP (or is it SIP with SIZZLE more to come)
I will commit to you to try to stick to the meat of SIP without the cheese of course
Please feel free to make this an interactive as possible!
Start with a Knowledge FoundationVOIP is a family of technologies, methodologies, communication protocols, and transmission techniques for the delivery of voice communications and multimedia sessions over IP networks, such as the Internet for example.
Session Initiation Protocol (SIP) is an signaling protocol for VOIP for creating, modifying, and terminating sessions with one or more participants of a VOIP call. Other well know signaling protocols are MGCP, H.323, SKINNY for examples
H.323 a call control element and signaling protocol that provides service to telephones or videophones. Such a device may provide or facilitate both basic services and supplementary services, such as call transfer, park, pick-up, and hold. IP-based PBX might be an H.323 Gatekeeper for example
Schools still in: More Basic VOIP TermsSkinny Client Control Protocol (SCCP) is a Cisco proprietary protocol used between Cisco Call Manager and Cisco VOIP phones. Referred to as Skinny only work with the SIP protocol and an example of a vendor VOIP only control protocol.
A Session Border Controller or SBC (IP to IP Gateway) is a device used in VOIP networks to allow control over the signaling and usually also the media streams involved in setting up, conducting, and tearing down calls. The are inserted into the signaling and/or media paths between calling and called parties in a VOIP call, predominantly those using the SIP, H.323, and MGCP call signaling protocols. Termed middle boxes between UAs and SIP servers.
A Media Gateway acts as a translation unit between disparate telecommunications networks such as the PSTN and Next Generation Networks . Media Gateways enable multimedia communications across these disparate networks over multiple transport protocols such as ATM and IP for example.
Almost DoneCODEC is a program capable of performing encoding and decoding on a digital data stream or signal. The word codec is actually just a combination of the words: compressor - decompressor. Common VOIP CODECS: G.711, G.729a, G.722 for example.
An IP PBX is a business telephone system designed to deliver voice or video over a data network and interoperate with the normal Public Switched Telephone Network (PSTN). Cisco Call Manager, Avaya, Microsoft, are a few examples.
A Soft Switch is a central device in a telecommunications network which connects telephone calls from one phone line to another, typically via the internet, entirely by means of software running on a general-purpose computer system that handles IP-to-IP phone calls. 2 types Class 4 and Class 5. SONUS and BroadSoft are examples.
Jitter is the undesired deviation of frequency of successive pulses in electronics and telecommunications. Jitter is a significant, and usually a undesired factor in the design of almost all communications links.
Basic Defined Elements in ActionCustomerPremiseFortisVoxeSBCIP PBXEthernetSwitch
Brief History of VOIP and the evolution of SIP with-in VOIP
It all started in 1995 and VocalTec
The history of VOIP shows that this technology started as far back as 1995 when a small company called VocalTec released, what was believed to be, the first internet phone software. This VOIP software was designed to run on a home PC and much like the PC phones used today, it utilized sound cards, microphones and speakers. The software was called "Internet Phone" and the hardware was called Audio Transceiver and used the H.323 protocol instead of the SIP protocol that is more dominant control protocol today. Anybody know what VocalTec is now most known for almost 20 years later?
Control Protocol EvolutionControl Protocols: Around since the mid-90sUsed to set up and break down VOIP sessions (Similar to the ISDN-PRI D-channel in a TDM environment)Types and different methodologies:H.323 - older ITU standard (hard to program or use)MGCP (Media Gateway Control Protocol) mostly used in Hosted VOIP or IP Centrex (never took off and Hosted has issues..)SIP (Session Initiation Protocol) - Has become the de facto control protocol (easy to program) SCCP helped / Beta vs VHS
Data Stream Protocol HistoryAfter a VOIP session is setup using a Control Protocol (SIP) then a Data Stream Protocol invades RTP (Realtime Transport Protocol) - Improves Quality of Service for VOIP data steams and used in VOIP today
2 RTP one way streams carry/enable the VOIP session (Similar to the ISDN-PRI B-channel in TDM Voice) RTCP (Realtime Transport Control Protocol) - used while the RTP Steam is running and piggy backs an RTP session to send summary reports back to sender
User Agents perform a series of SIP Commands to talkOnce again SIP is an Application Layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants, known as User Agents.
A series of SIP commands are used to accomplish the signaling tasks. Examples of these SIMPLE commands are:
INVITE: Invites a user to a callACK: Acknowledgement is used to facilitate reliable message exchange for INVITEs. BYE: Terminates a connection between users
THUS: SIP Session Call Flow a closer look2 versions of the same SIP session with the left version providing more of the details. The blue section shows the steps to setting up the session. The green section is the actual session using the two RTP streams and the Red section representing the breakdown stepsSetupSessionBreakdown
Deeper Dive on CODECsReferences compression software to COmpress and DECompress audio or video data streams to varying degree. Short for compress/decompress. CODECs can effect hardware and software (why there are many)Reduces the size of digital audio samples and video frames in order to:Speed up transmissionSave storage spaceSome CODECs discard bits that most people cannot hear or see for bit saving that effect quality levelsTrunk Calls will have typically have less compressed CODECs while higher compression is used in the LANs behind the IP PBX.
CODEC SpecificsG.711 is the default pulse code modulation (PCM) standard for Internet Protocol (IP) private branch exchange (PBX) vendors, as well as for the public switched telephone network (PSTN). G.711 digitizes analog voice signals producing output at 64 kilobits per second (Kbps). Since the late 1970's G.711 has been the defacto standard in the telephony world for voice encoding as we moved into the digital world with fully digital phone switches, and moved away from analog phone exchanges. Since the mid 90's as VoIP has rapidly taken over in the telephony world and G.711 has still remained as the codec of choice.
G.729 is an audio data compression algorithm for voice that compresses digital voice in packets of 10 milliseconds duration. Because of its low bandwidth requirements, G.729 is mostly used in Voice over Internet Protocol (VoIP) applications where bandwidth must be conserved.
G.722 HD Voice and HD Audio have become the latest buzzwords in the VoIP (Voice Over Internet Protocol) market in the last year. They are all words to describe the same thing - wideband audio that delivers voice calls using VoIP with audio quality that is greatly superior that of a regular landline or mobile phone call.
CODECsG.711 is the default CODEC for IP PBX vendors.and the PSTN
CODEC MisconceptionsG.711 is roughly 100K (87.2K) per call so a DS1 or 1.5m can handle 15 simultaneous calls.G.729 is roughly 40K (31.2K) per so a DS1 of 1.5m can of IP can handle 35 simultaneous calls
Obviously G.729 can save you money from the Vendor trunk side being less bandwidth is needed for more calls but if the design is off degradation, echo, dropped calls, etc can develop and VOIP/SIP can take the blame when really it just the CODEC. How? Remember when I said the PSTN is G.711? Scenario..
What about Jitter and SIP/VOIP?
JITTERJitter may be caused by electromagnetic interference (EMI) and crosstalk with carriers of other signals. Jitter can cause introduce clicks or other undesired effects in audio signals, and loss of transmitted data between network devices. The amount of tolerable jitter depends on the affected application.
Typically VOIP and SIP needs to operate with nothing more than 5ms of jitter at a max or again echo and degradation will occur.
Your SIP provider/vendor is very key to your success with SIP service being your providers network is what connects your SIP service for completion. Is your vendors network a shared or dedicated service? What is the latency on the network either layer 2 or layer3? Is a fiber based service or copper? Many more YOUR PROVIDER is KEY!
Ok - a little bit of bun
Month/YearPacket Delivery %Latency (ms)Jitter (ms)June-2011100.0038.870.03May -201